It is well-known that frequency modulation (FM) techniques can be used to synthesize harmonically-rich audio tones that are suitable for use in musical instruments (note: the term "frequency modulation" as used herein encompass any audio synthesis technique where the phase or frequency of a carrier signal is varied as a function of the content of a modulating signal).
Several such techniques are disclosed in U.S. Pat. No. 4,249,447, entitled "Tone production method for an electronic musical instrument." In each of the techniques disclosed in the '447 patent, the color of the synthesized output tone is at least partially modified by multiplying the modulating signal (say, sin(y)) by some feedback parameter (.beta.) then feeding back the resulting product (.beta.sin(y)) to be added to the phase of the carrier signal, thereby forming an updated (modulated) carrier phase value (y). The updated carrier phase value (y) is then input to a sinusoidal memory, which in response outputs the next value of the modulating signal (sin(y)).
The different techniques of the '447 patent can be used to synthesize audio tones with different characteristics (e.g., a square wave or a sine wave) by providing different types of feedback among the aforementioned basic components. However, audio synthesis circuits that implement the methods disclosed in the '447 patent are likely to introduce systematic inaccuracies in the phase signal (y) because, in each embodiment of the '447 patent, the signal being fed back to modify the current phase (y) is derived from a sinusoidal signal (e.g., sin(y)) output from a sinusoid memory/circuit.
This is because, in the art of audio synthesis, a sinusoid function is typically implemented as a logsin function followed by an addition and then a log-linear conversion. In this process, the current phase (y.sub.n) is input to a logsin function/memory, which outputs the log of sin(y) (i.e., logsin(y.sub.n)). The logsin signal (logsin(y.sub.n)) is then commonly added to a log-amplitude signal (log(A)) related to the envelope of the tone being synthesized. The resulting sum (log(A)+logsin(y.sub.n)) is then converted to a linear output signal (Asin(y.sub.n)) by a log-linear converter. These steps eliminate the need for an additional multiply, which is more costly than an addition and reduce the chance of computation overflow occurring. However, because information is lost in the logsin/addition/log-linear conversion process, the final result is less accurate (i.e., has fewer reliable lower-order bits) than if Asin(y.sub.n) were computed directly. In the FM audio synthesis systems employing methods of the '447 patent, these inaccuracies are accentuated by the fact that the resulting Asin(y.sub.n) signal is multiplied by a modulation index (.beta.), then that product is used to generate the phase value for the next audio synthesis cycle (y.sub.n+1). As a result, the current phase value is systematically thrown off during a synthesis operation.
Thus, there is a need for an audio synthesis circuit that does not feed back a sinusoid signal that is likely to have been log-linear converted. Ideally, such an audio synthesis circuit would instead feed back the current phase, compute a modulation factor from the current phase without using log-linear conversion, then form the next phase using that modulation factor. So that a wide variety of harmonics can be produced by this ideal system, the modulation factor should optionally be computed according to a function that differs from the sinusoidal function used to compute the output tones. The circuit should be structured so that this phase modulation operation is not applied to the output audio signal being synthesized.
For compatibility with legacy systems, i.e., audio synthesis systems that were designed around prior art audio synthesis circuits, these new audio synthesis circuits should provide a variety of legacy modes that are backward-compatible with audio synthesis modes provided by the prior art audio synthesis systems. For example one prior art system, the Yamaha OPL-3 audio synthesis chip, provides two modes, a 2-operator mode and a 4-operator mode, which allow a user to define complex audio voices by combining, respectively, 2 or 4 operators (each corresponding to an audio tone with different characteristics, such as frequency, envelope and harmonic content). In the OPL-3 chip, the association of operators and voices is fixed and no operator is used in more than one voice. For example, in the 4-op mode, the 36 operators might be allocated to six, four-operator voices and six, two-operator voices as follows:
______________________________________ Voice Operators ______________________________________ 1 1,2,3,4 2 5,6,7,8 3 9,10,11,12 4 13,14,15,16 5 17,18,19,20 6 21,22,23,24 7 25,26 8 27,28 9 29,30 10 31,32 11 33,34 12 35,36 ______________________________________
In the 2-op mode, two-operator voices are provided by subdividing the four-operator voices. I.e., the six, 4-op voices available in the 4-op mode become twelve 2-op voices in the 2-op mode. While this scheme allows a user to flexibly form 2-op or 4-op voices from the fixed set of operators, it also precludes a user who chooses to play a 2-op voice from playing the corresponding 4-op voice from which the 2-op voice was formed.
Thus, there is a need for mode selection circuitry for use in audio synthesis systems that provides backward-compatibility with prior art modes, while also providing an enhanced mode that does not preclude a user from simultaneously playing two and four operator voices, which, in the prior art would be corresponding and, therefore, non-playable.